I just want to play a very simple, straight forward note by giving my computer a certain frequency as an integer, and from there I can figure out how to make it play the note longer or shorter. It does not necessarily have to come out of the actual sound card - if it's generated and output by the internal speaker that's okay.
I looked at the midi libraries that java has included, and they are way more than what I want to do. This just needs to be very basic.
Look into JFugue -- it's really easy to do some basic stuff, and the capabilities are there if you want to expand later.
Player player = new Player();
player.play("A C# E");
This example constructs and plays an equal tempered scale.
As far as i know there is no way to do this without some boilerplate code in Java. The most simple API is probably provided by the Applet class (can be used by non-Applets as well) in the form of the static newAudioClip(URL url)-method. However this gives you just the ability to play predefined audio clips and you have very little control over the audio. If you just need to play audio from a small, predefined set of clips it might suffice (you could have a set of wav-files containing your notes and play them this way, AudioClip's can be looped if desired).
Other than that, both midi and sampled audio API's are much more powerful, but the flexibility comes at a price: You need considerably more code to set them up.
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I am attempting to write a very simple DAW in Java but am having trouble playing an audio clip in a sequence. I have looked into both the sampled and MIDI classes in Java Sound but what I really need is a hybrid of the two.
It seems that with the MIDI classes you cannot use a sequencer for example, to play your own audio clip.
I have attempted to write my own sequencer using scheduling to play a javax.sound.sampled.Clip in a sequence but the timings vary far too much. It is not really a viable option as it doesn't keep time.
Does anybody have any suggestions of how I could get around this?
I can attest that an audio mixing system combining aspects of MIDI and samples can be written in Java, as I wrote my own and it currently works with samples and a couple real-time synths that I also wrote.
The key is making the audio data of the samples available on a per-frame basis and a frame-counting command-processor/audio-mixer that both manages the execution of "commands," and collects and mixes the audio frame data. With 44100 fps, that's accuracy in the vicinity of 0.02 milliseconds. I can describe in more detail if requested.
Another way to go, probably saner, though I haven't done it personally, would be to make use of a Java bridge to a system such as Jack.
EDIT: Answering questions in comment (12/8/19).
Audio sample data in Java is usually either held in memory (Java uses Clip) or read from a .wav file. Because the individual frames are not exposed by Clip, I wrote an alternate, and use it to hold the data as signed floats ranging -1 to 1. Signed floats are a common way to hold audio data that we are going to perform multiple operations upon.
For playback of .wav audio, Java combines reading the data with AudioInputStream and outputting with SourceDataLine. Your system will have to sit in the middle, intercepting the AudioInputStream, convert to PCM float frames, and counting the frames as you go.
A number of sources or tracks can be processed at the same time, and merged (simple addition of the normalized floats) to a single signal. This signal can be converted back to bytes and sent out for playback via a single SourceDataLine.
Counting output frames from an arbitrary 0th frame from the single SourceDataLine will help with keeping constituent incoming tracks coordinated, and will provide the frame number reference used to schedule any additional commands that you wish to execute prior to that frame being output (e.g., changing a volume/pan of a source, or a setting on a synth).
My personal alternate to a Clip is very similar to AudioCue which you are welcome to inspect and use. The main difference is that for better or worse, I'm processing everything one frame at a time in my system, and AudioCue and its "Mixer" process buffer loads. I've had several very credible people criticize my personal per-frame system as inefficient, so when I made the public API for AudioCue, I bowed to that preconception. [There are ways to add buffering to a per-frame system to recapture that efficiency, and per-frame makes scheduling simpler. So I'm sticking with my per-frame logical scheme.]
No, you can't use a sequencer to play your own clips directly.
In the MIDI world, you have to deal with samples, instruments, and soundbanks.
Very quickly, a sample is the audio data + informations such as looping points, note range covered by the sample, base volume and envelopes, etc.
An instrument is a set of samples, and a soundbank contain a set of instruments.
If you want to use your own sounds to play some music, you must make a soundbank out of them.
You will also need to use another implementation than the default provided by Java, because that default only read soundbanks in a proprietary format, which is gone since at least 15 and perhaps even 20 years.
Back in 2008-2009, there existed for example Gervill. It was able to read SF2 and DLS soundbanks. SF2 and DLS are two popular soundbank formats, several programs exist in the market, free or paid, to edit them.
If you want to go from the other way round, starting with sampled, that's also exact as you ahve noticed, you can't rely on timers, task schedule, Thread.sleep and the like to have enough precision.
The best precision you can achieve by using those is around 10ms, what's of course far too few to be acceptable for music.
The usual way to go here is to generate the audio of your music by mixing your audio clips yourself into the final clip. So you can achieve frame precision.
In fact that's very roughly what does a MIDI synthesizer.
I am a musician/singer/songwriter,
I was hoping someone might know of information already out that does some if not all of what I'm trying to achieve.
I record song ideas into raw digital wav files using only my voice to emulate instruments ( vocal melody, bass, guitar, drums, etc.) into a song structure (verse, chorus, bridge).
I was hoping that java/fft could be used to slice each mili-second into an array that could be broken down into notes and riffs that I am singing.
Here is a list of some of the steps I see that need to be done with my wav files.
Find out the note that I'm singing. The software would take each note and nudge it into the nearest "true note" (a4=440hz).
It would take the notes and find out which key or possible keys the song may be in.
From a very large database of real songs, the software would make chord suggestions and placement suggestions depending on the genre the song is in.
It would take the riffs ( any sequence of more than 3 notes done more than 3 times in a song) and create loops with a drop down box of alternative voiceings and randomizing.
There is much more, but this should show you the basics of what I’m trying to do.
If there aren’t programs already written that already do all or part of this, would it be possible for me to write a program that uses java and fft to slice every millisecond into an array to determine notes?
I have read some java/fft material and it is way over my head mostly (I have studied a little java) but I was hoping someone might be able to lead me in the right direction.
Android provides a default of 15 steps for its sound systems which you can access through Audio Manager. However, I would like to have finer control.
One method of doing so seems to be altering specific files within the Android system to divide the sound levels even further then default. I would like to programmatically achieve the same effect using Java.
Fine volume control is an example of the app being able to divide the sound levels into one hundred distinct intervals. How do I achieve this?
One way, in Java, to get very precise volume adjustment is to access the PCM data directly and multiply it by some factor, usually from 0 up to 1. Another is to try and access the line's volume control, if it has one. I've given up trying to do the latter. The precision is okay in terms of amplitude, but the timing is terrible. One can only have one volume change per audio buffer read.
To access the PCM data directly, one has to iterate through the audio read buffer, translate the bytes into PCM, perform the multiplication then translate back to bytes. But this gives you per-frame control, so very smooth and fast fades can be made.
EDIT: To do this in Java, first check out the sample code snippet at the start of this java tutorial link, in particular, the section with the comment
// Here, do something useful with the audio data that's now in the audioBytes array...
There are several StackOverflow questions that show code for the math to convert audio bytes to PCM and back, using Java. Should not be hard to uncover with a search.
Pretty late to the party, but I'm currently trying to solve this issue as well. IF you are making your own media player app and are running an instance of a MediaPlayer, then you can use the function setVolume(leftScalar, rightScalar) where leftScalar and rightScalar are floats in the range of 0.0 to 1.0. representing logarithmic scale volume for each respective ear.
HOWEVER, this means that you must have a reference to the currently active MediaPlayer instance. If you are making a music app, no biggie. If you're trying to run a background service that allows users to give higher precision over all media output, I'm not sure how to use this in that scenario.
Hope this helps.
So I can get an array of all the bytes of a wav file, I just want to know how I can decode the raw sound data to something I can use to tell when the singer is speaking/his beat (I don't know the proper musical terms, sorry)
If there is an API or tutorial out there that someone could link me to, that would be swell since I can't seem to find anything good.
Will you know this beat in advance? If so, you could cross correlate the two signals and the highest peak in this output would correspond to the time delay.
Other than that, depending on the sound before the beat starts, you could convert to frequency domain (via FFT) and have a look at what frequencies are present and see whether there's a significant change when the beat begins.
Some examples/extra detail would help.
If you're trying to detect the tempo of said beat, please ignore everything most of what I've said.
In general, detecting "the instances when something beats" in a wave file is not as one may imagine at the first thought.
A possible first step is to transform your .wav into a so-called "spectrogram."
I don't think Java has a dedicated API for this purpose, but googling "java spectrogram" would give you a number of third-party examples.
I also found this question might be relevant.
P.S. I'm not a specialist in signal processing, so corrections are welcome.
So basically I want this to get the range of 60 - 150 Hz which is the general area for bass that lies in a song. Whenever it is in this range I want it do a function, and only it the range, my problem is I have tried to look up the functions needed to do so but with no luck, if one could show me here or a good article or explanation on this it will be great! I appreciate all the help and I will continue looking on my own. If more explanation is needed I can provide whatever information that is needed!
Austin.
UPDATE: I simplified an algorithm here:
User selects the song they want
Song loads onto player
Function scans song and finds the lower frequencies throughout the song and the output is a pattern.
Step 1) Do a fast fourier transform: http://en.wikipedia.org/wiki/Fast_Fourier_transform
An FFT takes a piece of sound and transforms it into the frequency/time domain - as in, which frequencies are playing and how intensely and during what parts of the sound. This is a useful mathematical operation that relies upon the property that all sound, no matter how complex, can be fundamentally constructed out of one or more sine waves of different frequencies and amplitudes.
If you've ever looked at a spectrogram, for example in foobar2000, it is implemented using FFT:
I suggest instead of trying to implement FFT yourself you find a library that is well tested and fast, such as http://en.wikipedia.org/wiki/FFTW which is written in C
Step 2) Now that you've FFTed the part of the sound that the user is listening to, you can simply inspect the frequency bins and do whatever you want! Although detecting bass kicks is not as simple as 'is this frequency bin a high value?' because then you may mistake bass lines for bass kicks. You may need to do further testing and research to get it to work juuust right.
EDIT: Delyan suggests http://www.clear.rice.edu/elec301/Projects01/beat_sync/beatalgo.html and it looks pretty good.